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peerstreamer-src / Tools / janus_conf / janus.plugin.streaming.cfg @ c81c126e

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; [stream-name]
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; type = rtp|live|ondemand|rtsp
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;        rtp = stream originated by an external tool (e.g., gstreamer or
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;              ffmpeg) and sent to the plugin via RTP
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;        live = local file streamed live to multiple listeners
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;               (multiple listeners = same streaming context)
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;        ondemand = local file streamed on-demand to a single listener
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;                   (multiple listeners = different streaming contexts)
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;        rtsp = stream originated by an external RTSP feed (only
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;               available if libcurl support was compiled)
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; id = <unique numeric ID> (if missing, a random one will be generated)
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; description = This is my awesome stream
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; is_private = yes|no (private streams don't appear when you do a 'list'
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;			request)
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; secret = <optional password needed for manipulating (e.g., destroying
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;			or enabling/disabling) the stream>
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; pin = <optional password needed for watching the stream>
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; filename = path to the local file to stream (only for live/ondemand)
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; audio = yes|no (do/don't stream audio)
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; video = yes|no (do/don't stream video)
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;    The following options are only valid for the 'rtp' type:
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; data = yes|no (do/don't stream text via datachannels)
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; audioport = local port for receiving audio frames
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; audiomcast = multicast group port for receiving audio frames, if any
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; audioiface = network interface or IP address to bind to, if any (binds to all otherwise)
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; audiopt = <audio RTP payload type> (e.g., 111)
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; audiortpmap = RTP map of the audio codec (e.g., opus/48000/2)
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; videoport = local port for receiving video frames
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; videomcast = multicast group port for receiving video frames, if any
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; videoiface = network interface or IP address to bind to, if any (binds to all otherwise)
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; videopt = <video RTP payload type> (e.g., 100)
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; videortpmap = RTP map of the video codec (e.g., VP8/90000)
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; videobufferkf = yes|no (whether the plugin should store the latest
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;		keyframe and send it immediately for new viewers, EXPERIMENTAL)
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; videosimulcast = yes|no (do|don't enable video simulcasting)
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; videoport2 = second local port for receiving video frames (only for rtp, and simulcasting)
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; videoport3 = third local port for receiving video frames (only for rtp, and simulcasting)
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; dataport = local port for receiving data messages to relay
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; dataiface = network interface or IP address to bind to, if any (binds to all otherwise)
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; databuffermsg = yes|no (whether the plugin should store the latest
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;		message and send it immediately for new viewers)
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;
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; The following options are only valid for the 'rstp' type:
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; url = RTSP stream URL (only for restreaming RTSP)
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; rtsp_user = RTSP authorization username (only if type=rtsp)
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; rtsp_pwd = RTSP authorization password (only if type=rtsp)
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; rtspiface = network interface or IP address to bind to, if any (binds to all otherwise), when receiving RTSP streams
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;
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; Notice that, for 'rtsp' mountpoints, normally the plugin uses the exact
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; SDP rtpmap and fmtp attributes the remote camera or RTSP server sent.
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; In case the values set remotely are known to conflict with WebRTC viewers,
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; you can override both using the settings introduced above.
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;
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; To test the [gstreamer-sample] example, check the test_gstreamer.sh
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; script in the plugins/streams folder. To test the live and on-demand
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; audio file streams, instead, the install.sh installation script
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; automatically downloads a couple of files (radio.alaw, music.mulaw)
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; to the plugins/streams folder. 
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[general]
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;admin_key = supersecret		; If set, mountpoints can be created via API
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								; only if this key is provided in the request
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;events = no					; Whether events should be sent to event
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								; handlers (default is yes)
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;[gstreamer-sample]
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;type = rtp
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;id = 1
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;description = Opus/VP8 live stream coming from gstreamer
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;audio = yes
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;video = yes
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;audioport = 5002
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;audiopt = 111
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;audiortpmap = opus/48000/2
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;videoport = 5004
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;videopt = 100
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;videortpmap = VP8/90000
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;secret = adminpwd
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;
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;[file-live-sample]
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;type = live
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;id = 2
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;description = a-law file source (radio broadcast)
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;filename = /home/baldo/src/peerstreamer-src/Tools/janus1/share/janus/streams/radio.alaw		; See install.sh
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;audio = yes
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;video = no
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;secret = adminpwd
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;
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;[file-ondemand-sample]
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;type = ondemand
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;id = 3
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;description = mu-law file source (music)
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;filename = /home/baldo/src/peerstreamer-src/Tools/janus1/share/janus/streams/music.mulaw	; See install.sh
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;audio = yes
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;video = no
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;secret = adminpwd
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;
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; Firefox Nightly supports H.264 through Cisco's OpenH264 plugin. The only
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; supported profile is the baseline one. This is an example of how to create
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; a H.264 mountpoint: you can feed it an x264enc+rtph264pay pipeline in
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; gstreamer.
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;
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;[h264-sample]
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;type = rtp
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;id = 10
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;description = H.264 live stream coming from gstreamer
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;audio = no
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;video = yes
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;videoport = 8004
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;videopt = 126
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;videortpmap = H264/90000
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;videofmtp = profile-level-id=42e01f\;packetization-mode=1
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;
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; This is a sample configuration for Opus/VP8 multicast streams
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;
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;[gstreamer-multicast]
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;type = rtp
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;id = 20
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;description = Opus/VP8 live multicast stream coming from gstreamer 
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;audio = yes
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;video = yes
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;audioport = 5002
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;audiomcast = 232.3.4.5
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;audiopt = 111
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;audiortpmap = opus/48000/2
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;videoport = 5004
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;videomcast = 232.3.4.5
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;videopt = 100
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;videortpmap = VP8/90000
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;
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; This is a sample configuration for an RTSP stream: you can specify
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; the url to connect to and whether or not authentication is needed
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; using the url/rtsp_user/rtsp_pwd settings (but notice that digest
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; authentication will only work if you installed libcurl >= 7.45.0)
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; NOTE WELL: the plugin does NOT transcode, so the RTSP stream MUST be
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; in a format the browser can digest (e.g., VP8 or H.264 baseline for video)
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; Again, you can override rtpmap and/or fmtp, if needed
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;
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;[rtsp-test]
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;type = rtsp
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;id = 99
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;description = RTSP Test
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;audio = no
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;video = yes
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;url=rtsp://127.0.0.1:8554/unicast
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;rtsp_user=username
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;rtsp_pwd=password